[asterisk-bugs] [JIRA] (ASTERISK-26386) chan_sip: ACK being sent is referencing the contact on the Request-URI instead of To

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26386?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-26386:
-----------------------------------

    Assignee: Rajiv Duggal  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

The most recent release of 13 is Asterisk 13.11.2. I'm unfamiliar with how to update from FreePBX, you would need to look at their documentation for doing so. I'm putting this back into feedback until we know if it remains an issue with a recent version as well.

> chan_sip: ACK being sent is referencing the contact on the Request-URI instead of To
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26386
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26386
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/Interoperability
>    Affects Versions: 13.2.0
>         Environment: 64bit Linux
>            Reporter: Rajiv Duggal
>            Assignee: Rajiv Duggal
>         Attachments: sip.log
>
>
> per RFC protocol (https://tools.ietf.org/html/rfc3261#page-129) 
> the generated ACK should use the TO from the response being acknowledged
> This is causing call disconnects and continuous invites from some SIP providers who receive a contact from upstream which is not identical to the TO part of the header.
> This can be replicated for phone # 15033277805.
> received header per below
> {noformat}
> 17:03:36.856392 IP (tos 0x0, ttl 53, id 37874, offset 0, flags [none], proto UDP (17), length 1057)
> 65.254.44.194.sip > 10.2.1.40.sip-tls: [udp sum ok] SIP, length: 1029
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 100.32.192.11:5061;received=100.32.192.11;branch=z9hG4bK3da029f8;rport=5061
> From: <sip:8053881711 at gw1.sip.us:5061>;tag=as28725f49
> To: <sip:15033277805 at gw1.sip.us:5060>;tag=gK0ce47da4
> Call-ID: 0e760a970d24e31e22afca333f10989e at gw1.sip.us
> CSeq: 103 INVITE
> Record-Route: <sip:67.231.0.87:5060;lr=on;ftag=as28725f49>
> Record-Route: <sip:65.254.44.194:5060;lr=on;ftag=as28725f49;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAADo1MDYx;proxy_media=yes;dlgcor=018.b1f>
> Accept: application/sdp
> Contact: <sip:+15033277805 at 192.168.16.8:5060>
> Allow: INVITE,ACK,CANCEL,BYE,REFER,PRACK,OPTIONS
> Supported: timer
> Session-Expires: 1800;refresher=uas
> Content-Length: 258
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
> {noformat}
> Note contact has a +prefixed above and below ACK generated by the system.
> {noformat}
> 17:03:36.857085 IP (tos 0x60, ttl 64, id 26156, offset 0, flags [none], proto UDP (17), length 629)
> 10.2.1.40.sip-tls > 65.254.44.194.sip: [udp sum ok] SIP, length: 601
> ACK sip:+15033277805 at 192.168.16.8:5060 SIP/2.0
> Via: SIP/2.0/UDP 100.32.192.11:5061;branch=z9hG4bK148ab810;rport
> Route: <sip:65.254.44.194:5060;lr=on;ftag=as28725f49;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAADo1MDYx
> ;proxy_media=yes;dlgcor=018.b1f>,<sip:67.231.0.87:5060;lr=on;ftag=as28725f49>
> Max-Forwards: 70
> From: <sip:8053881711 at gw1.sip.us:5061>;tag=as28725f49
> To: <sip:15033277805 at gw1.sip.us:5060>;tag=gK0ce47da4
> Contact: <sip:8053881711 at 100.32.192.11:5061>
> Call-ID: 0e760a970d24e31e22afca333f10989e at gw1.sip.us
> CSeq: 103 ACK
> User-Agent: FPBX-AsteriskNOW-12.0.76.4(13.2.0)
> Content-Length: 0
> {noformat}
> this is causing issues with some SIP providers. Complete log is attached.



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